KR20050053919A - System and method servicing number portability in an integration network of the voice over internet protocol network and the public switched telephone network - Google Patents

Abstract

Disclosed are a VoIP and public telephone integrated service system and method for providing number portability. The VoIP and public telephone integrated service system includes a number portability service from a public telephone network to a VoIP network, a number portability service from a VoIP network to a public telephone network, or a number portability between VoIP networks in a soft switch interworking a VoIP network and a public telephone network. In particular, in the number portability service between VoIP networks, the service can be directly routed from the SIP proxy server to the SIP terminal, thereby improving performance.

Description

System and method servicing number portability in an integration network of the voice over internet protocol network and the public switched telephone network

The present invention relates to an Internet-based integrated network interworking voice over internet protocol (VoIP) networks and public telephone networks, and more particularly, to an integrated service system and method for providing number portability in the Internet-based integrated network.

Number portability service is provided in public telephone network or mobile communication network. In the public telephone network, it is classified and serviced as Local Number Portability (LNP), and in the mobile communication network, it is classified and serviced as Mobile Number Portability (MNP). Typical implementations of local phone number portability include Remote Call Forwarding (RCF), Query on Release (QoR), and All Call Query (ACQ). Typical implementations of mobile number portability include a Query on HLR Release (QoRH), Terminating Call Query on Digit analysis (TQoD), and Original Call Query on Digit analysis (OQoD).

The typical mobility service method is as follows. In the RCF method, the receiving party operator exchange detects the fact that the user has changed the communication provider and transmits the detected new destination operator to the new destination exchange. In this manner, since all calls are delivered to the existing operator exchange, effective routing is difficult, not only waste of number resources but also a long time for call processing. In addition, in the QoR method, when a call occurs, a call release message is sent from the originating exchange to the destination exchange, and a method of checking whether the corresponding number has been moved in the destination exchange is used. At this time, if the number is transferred, the call is released, and the cause value of the release message is indicated and sent to the calling party, and the calling party connects to the LNP database to obtain information to be routed to the new destination. In this manner, there is a problem in that a call delay occurs due to a database connection in the case of a subscriber who has changed the service provider.

Meanwhile, with the proliferation of the Internet, number portability services are emerging in VoIP networks. However, when using a conventional method such as QoR when providing a number porting service from a VoIP network to a VoIP network, a SIP (Session Initiation Protocol) server does not directly transfer a call from an existing destination to a new destination, but instead sends a call to an originating exchange. Since the release is required, there is a problem in that performance is degraded as pointed out above. In addition, conventionally, since the number portability service is not provided for a softswitch interworking in a public telephone network and a VoIP network, it is necessary to actively cope with an increase in the number portability service demand due to various increase of network operators and service providers. have.

Accordingly, the present invention has been made to solve the above problems, the technical problem to be achieved by the present invention, in the Internet-based integrated network in which the VoIP network and the public telephone network interworked, the number portability service through the soft switch and SIP The present invention provides a VoIP and public telephone integrated service system and method capable of performing call processing for a number portability service through a proxy server.

The VoIP and public telephone integrated service system according to the present invention for achieving the above technical problem, in the Internet-based integrated network in which the VoIP network and the public telephone network interworked, extracts the destination terminal number in response to the call request of the calling terminal; 1 an outgoing network that transmits an outgoing call to a called terminal through a soft switch and resends the outgoing call to a changed address or number in response to call release information; In response to the outgoing call, when the called terminal number is changed to a shifted number, transmitting the call release information to the first soft switch, or directly retransmitting the outgoing call to a new called terminal having the shifted number. Pre-mobile incoming network; And a new incoming network that performs call connection routing between the calling terminal and the new called terminal in response to the resent outgoing call. The originating network operates in a public telephone network having a PSTN switch or in a VoIP network having a SIP proxy server.

If the pre-mobile destination network operates in a public telephone network having a PSTN exchanger and the new destination network operates in a VoIP network having a SIP proxy server, the pre-mobile destination network moves through the second softswitch. The call release information including the received number is transmitted, and the calling network retransmits the calling call to the new called terminal having the moved number in response to the call release information. Here, the shifted number is a new SIP terminal address in which the called terminal number is changed.

When the pre-mobile destination network operates in a VoIP network having a SIP proxy server, and the new destination network operates in a public telephone network having a PSTN switch, the pre-mobile destination network is moved through the SIP proxy server. The call release information including the number is transmitted, and the calling network retransmits the calling call to the new called terminal having the moved number in response to the call release information. Here, the shifted number is a new public telephone terminal number in which the called terminal number is changed.

When both the pre-mobile and the new destination network operate in a VoIP network having a SIP proxy server, the pre-mobile destination network directly retransmits the outgoing call to the new destination terminal having the ported number. It features. Here, the shifted number is a new SIP terminal address in which the called terminal number is changed.

The VoIP and public telephone integrated service method according to the present invention for achieving the above another technical problem, in the Internet-based integrated network in which the VoIP network and the public telephone network interworked, in the calling network, in response to the call request of the calling terminal Extracting a number and transmitting an outgoing call to the called terminal through the first soft switch; Transmitting a call release information to the first soft switch when the called terminal number is changed to the moved number in response to the outgoing call in the pre-moving called network; Retransmitting the outgoing call to the changed address or number in response to the call release information in the outgoing network; And in the new destination network, performing call connection routing between the originating terminal and the new destination terminal in response to the retransmitted outgoing call. In the VoIP and public telephone integrated service method, when the destination terminal number is changed to a ported number in response to the outgoing call, the originating call is directly transferred to a new destination terminal having the ported number. And retransmitting.

In order to fully understand the present invention, the operational advantages of the present invention, and the objects achieved by the practice of the present invention, reference should be made to the accompanying drawings which illustrate preferred embodiments of the present invention and the contents described in the accompanying drawings.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. Like reference numerals in the drawings denote like elements.

In the international standardization forum related to open services, softswitch call processing requires the translation and routing of numbers and the provision of services with various communication functions for various call types for each network type. Soft switch development based on open technology means providing flexibility in the network and allowing a number of service developers outside the network to access the communication functions existing in the network. The present invention proposes a number portability service providing method in a public telephone network and a VoIP network, which are the current development trends of a network. In particular, in the present invention, the number portability service when the destination network is a public telephone network and the new destination network is a VoIP network, the number portability service when the destination network is a VoIP network and the new destination network is a public telephone network, and a destination before mobile We propose a method for providing number portability service when both the network and the new destination network are VoIP networks.

1 is a block diagram of a VoIP and public telephone integrated service system according to an embodiment of the present invention for providing a number portability service when the destination network before the mobile is a public telephone network and the new destination network is a VoIP network. FIG. 2 is a block diagram of an integrated VoIP and public telephone service system according to an embodiment of the present invention for providing number portability service when the destination network before the mobile network is a VoIP network and the new destination network is a public telephone network. 3 is a block diagram of a VoIP and public telephone integrated service system according to an embodiment of the present invention for providing number portability service when the destination network before the mobile network is a VoIP network and the new destination network is a VoIP network.

1 to 3, a VoIP and public telephone integrated service system according to an embodiment of the present invention includes an outgoing network 110, a pre-mobile incoming network 130, and a new incoming network 150. They are connected to the Internet-based converged network 120 where the VoIP network and the public switched telephone network (PSTN) work together.

The calling network 110 extracts the called terminal number in response to the call request of the calling terminal 111 to the called terminal (131 in FIG. 1, 151 in FIG. 1 and FIG. 3) through the first soft switch 114. It sends an IIA (Initial Address Message) and resends the IAM to the changed address or number in response to call release information (REL). 1 to 3, the outgoing network 110 is shown to operate in a public telephone network having a LEX (Local Exchange) PSTN switch 112, but may also operate in a VoIP network having a SIP proxy server.

1 to 3, the soft switches 114 and 134 and the media gateways 113 and 133 serve as core nodes for directly connecting the VoIP network and the PSTN network, and between the two nodes, the MEGACO protocol. The standard interface in operation is used. H.248 may be used as the interface between the two nodes. The soft switches 114 and 134 provide call control functions such as connection and control of calls, number translation and routing functions, and number portability functions, and media gateways 113 and 133 control and management functions. The media gateways 113 and 133 are in charge of the exchange transmission under the control of the soft switches 114 and 134, and are a path control function for transmitting subscriber terminal information and mutual media conversion between the pulse code conversion stream of the PSTN and the Internet packet. And so on. The LEX PSTN exchanges 112 and 132 provide a function of delivering a message to a subscriber station connected to the PSTN network. 1 to 3 illustrate a call processing form between different soft switches 114 and 134. A session initiation protocol-telephone (SIP-T) or a bearer independent call control (BICC) protocol is used between the soft switches 114 and 134. Can be used. Interface and trunk line management between the LEX PSTN exchangers 112 and 132 and the soft switches 114 and 134 are performed by an ISDN user part (ISP). ISDN (Integrated Services Digital Network) is a comprehensive information network.

Referring to FIG. 1, the pre-mobile terminating network 130 operates in a public telephone network having a PSTN exchanger 132, and the new terminating network 150 includes a SIP proxy server 152. In the case of operating in a VoIP network, if the destination network 130 before the movement determines that the number has been translated as a result of number translation in response to the outgoing call (IAM) from the originating network 110, the second soft switch 134. Call release information (REL) including the shifted number is transmitted through e). Accordingly, in response to the call release information (REL), the first soft switch 114 provided in the calling network 110 searches the intelligent network database to extract the moved number, and the new called network 150 is a VoIP network. In this case, the first soft switch 114 retransmits the outgoing call IAM to the new SIP destination terminal 151 having the ported number. In the case of Fig. 1, the moved number corresponds to a new SIP terminal address of which the called terminal number is changed. The new called network 150 performs call connection routing between the calling terminal 111 and the new called terminal 151 in response to the retransmitted outgoing call (IAM).

Referring to FIG. 2, the pre-mobile destination network 130 operates in a VoIP network including a SIP proxy server 152, and the new destination network 150 operates in a public telephone network including a PSTN exchanger 132. In the case of doing so, the destination network 130 before the mobile device determines whether the number has been moved as a result of the number translation of the SIP proxy server 152 in response to the outgoing call (IAM) from the originating network 110, and the location database. (Location DB) 153 is searched to determine whether the ported number belongs to a VoIP network or a public telephone network. If it is determined that the transferred number is a number belonging to a public telephone network, the before-moving destination network 130 transmits call release information 300 (MOVED) indicating that the call has been moved through the SIP proxy server 152. Accordingly, the first soft switch 114 provided in the calling network 110 searches for an intelligent network database in response to the call release information {300 (MOVED)}, extracts the moved number, and receives the new incoming network 150. In this public telephone network, the first soft switch 114 retransmits the outgoing call IAM to the new public telephone terminating terminal 131 having the ported number. In the case of Fig. 2, the shifted number corresponds to a new public telephone terminal number in which the called terminal number is changed. The new called network 150 performs call connection routing between the calling terminal 111 and the new called terminal 131 in response to the retransmitted outgoing call (IAM).

Referring to FIG. 3, when both the pre-mobile terminating network 130 and the new terminating network 150 operate in a VoIP network including SIP proxy servers 152 and 162, the pre-mobile terminating network 130 In response to the outgoing call (IAM) from the originating network 110, and determines that the called terminal number has been changed to the ported number through the SIP proxy server 152, the new called terminal having the ported number. Directly resend the originating call (IAM) to 161. In the case of Figure 3, since the call processing from the existing incoming SIP proxy server 152 to the new incoming SIP proxy server 162, the number translation process is unnecessary and the new incoming network ( Since the SIP terminal address 161 of 150 is allocated, even if the QoR number portability scheme is used, the process of releasing and transmitting a call to the originating network 110 is unnecessary and omitted. The new called network 150 performs call connection routing between the calling terminal 111 and the new called terminal 131 in response to the retransmitted outgoing call (IAM).

Hereinafter will be described in more detail the operation of the VoIP and public telephone integrated service system according to an embodiment of the present invention.

4 is a flowchart illustrating the operation of the VoIP and public telephone integrated service system of FIG. Referring to FIG. 4, the call release information including the "Number Changed" value indicating that the number has been moved to the calling party's first soft switch 114 by checking that the number has been moved after translating the number from the destination network 130 before moving ( A process of transmitting the REL) and a process of the caller-side first soft switch 114 receiving the call release information REL are illustrated. When the VoIP network and users are expanded, several softswitches must exist for efficient call processing. Therefore, the number portability service function must be created in the softswitch. First, the outgoing call (IAM) requested from the subscriber originating terminal 111 is transferred to the number translation block in the ISUP, and is transferred to the called party's soft switch 134 determined after the number translation (S410). The called party soft switch 134 receives the call release information REL including the number value ("Number Changed") moved from the ISUP and transmits it to the calling party soft switch 114 (S420). The originating network 110 receives the call release information REL, and if the call release information REL includes the shifted number value, transfers the call to an intelligent network (SCP: Switch Control Point) through the ISUP, and then to the intelligent network database. Search for a new address at step S430. The originating network 110 requests a number search (IN) from the intelligent network database, receives a new connection address, transfers it to the ISUP, and retransmits the outgoing call (IAM) to the new destination network 150 to establish a call connection. Try. The SIP proxy server 152 of the new destination network 150 performs call connection routing between the originating terminal 111 and the new destination terminal 131 in response to the retransmitted outgoing call (IAM).

5 is a flowchart illustrating an operation of the VoIP and public telephone integrated service system of FIGS. 2 and 3. Referring to FIG. 5, when the number portability service of the soft switches 114 and 134 interworking the VoIP network and the public telephone network is provided, the destination network 130 before the movement is the VoIP network, and the new destination network 150 is the public telephone network or In the case of a VoIP network, an operation process of the SIP proxy server 152 in the destination mobile network 130 for performing efficient call control is illustrated. After receiving the call request from the originating first soft switch 114, the SIP proxy server 152 of the destination network 130 retrieves the call information from the location database 153 connected to the SIP proxy server 152. (S510, S520). At this time, if the requested call is a moved number, the new destination number is retrieved from the location database 153 (S530 and S550). At this time, if there is a direct redirection address, since the new destination network 150 is a VoIP network, the SIP newly changed to the direct new destination network 150 without having to release the call to the first soft switch 114 of the calling party. Forward the call to the address (S560, S570). If the requested call is not the moved number, call connection routing is performed in the same manner as the call processing of the existing SIP proxy server 152 (S540). If the requested call is a ported number, but there is no direct redirection address, since the new destination network 150 is a public telephone network, 300 indicating that the number has been moved to the calling party first soft switch 114 is moved. After transmitting the data (308), the calling party first soft switch (114) re-searches the transferred number through the intelligent network database, and then sends the call to the new destination network (150) operating in the public telephone network according to the processing result. Resend (IAM).

As described above, the VoIP and public telephone integrated service system according to an embodiment of the present invention, the number portability service from the public telephone network to the VoIP network in the soft switch (114, 134) interworking the VoIP network and the public telephone network , A number portability service from a VoIP network to a public telephone network, or a number portability service between VoIP networks, and in particular, in the number portability service between VoIP networks, the SIP proxy server 152 can directly route to the SIP terminal 161. Performance is improved.

The functions used in the systems and methods disclosed herein can be embodied as computer readable code on a computer readable recording medium. The computer-readable recording medium includes all kinds of recording devices in which data that can be read by a computer system is stored. Examples of computer-readable recording media include ROM, RAM, CD-ROM, magnetic tape, floppy disk, optical data storage, and the like, and may also be implemented in the form of a carrier wave (for example, transmission over the Internet). do. The computer readable recording medium can also be distributed over network coupled computer systems so that the computer readable code is stored and executed in a distributed fashion.

As described above, optimal embodiments have been disclosed in the drawings and the specification. Although specific terms have been used herein, they are used only for the purpose of describing the present invention and are not intended to limit the scope of the invention as defined in the claims or the claims. Therefore, those skilled in the art will understand that various modifications and equivalent other embodiments are possible from this. Therefore, the true technical protection scope of the present invention will be defined by the technical spirit of the appended claims.

VoIP and public telephone integrated service system according to the present invention, in the Internet-based integrated network interworking VoIP network and public telephone network, call processing for number portability service through softswitch and number portability service through SIP proxy server Do. In addition, even when the QoR method is used in the number portability service between VoIP networks, an unnecessary step of newly requesting a call to a new destination after transmitting the released call to the originating exchange is omitted, thereby increasing the efficiency of the number portability service.

1 is a block diagram of a VoIP and public telephone integrated service system according to an embodiment of the present invention for providing a number portability service when the destination network before the mobile is a public telephone network and the new destination network is a VoIP network.

FIG. 2 is a block diagram of an integrated VoIP and public telephone service system according to an embodiment of the present invention for providing number portability service when the destination network before the mobile network is a VoIP network and the new destination network is a public telephone network.

3 is a block diagram of a VoIP and public telephone integrated service system according to an embodiment of the present invention for providing number portability service when the destination network before the mobile network is a VoIP network and the new destination network is a VoIP network.

 4 is a flowchart illustrating the operation of the VoIP and public telephone integrated service system of FIG.

5 is a flowchart illustrating an operation of the VoIP and public telephone integrated service system of FIGS. 2 and 3.

<Description of Major Symbols in Drawing>

114, 134: Soft switch 112, 132: PSTN exchanger

113, 133: Media Gateways 152, 162: SIP Proxy Server

Claims (17)

In the Internet-based integrated network in which a VoIP network and a public telephone network are interworked, An originating network extracting a called terminal number in response to a call request of a calling terminal, transmitting an outgoing call to a called terminal through a first soft switch, and resending the calling call to a changed address or number in response to call release information; In response to the outgoing call, when the called terminal number is changed to a shifted number, transmitting the call release information to the first soft switch, or directly retransmitting the outgoing call to a new called terminal having the shifted number. Pre-mobile incoming network; And And a new terminating network for performing call connection routing between the originating terminal and the new terminating terminal in response to the retransmitted outgoing call. The method of claim 1, wherein the source network, VoIP and public telephone integrated service system operating in a public telephone network having a PSTN switch. The method of claim 1, wherein the source network, A VoIP and public telephone integrated service system, which operates in a VoIP network having a SIP proxy server. According to claim 1, The incoming network before the movement, When the new destination network is operated on a VoIP network having a SIP proxy server, the before-mobile destination network includes the ported number via a second softswitch. And transmitting the release information, wherein the calling network retransmits the calling call to a new called terminal having the ported number in response to the call release information. According to claim 1, The incoming network before the movement, When the new destination network operates on a public telephone network having a PSTN exchange, and the before-mobile destination network includes the ported number through a SIP proxy server. And the originating network retransmits the originating call to a new called terminal having the ported number in response to the call release information. The method of claim 5, wherein the shifted number is, VoIP and public telephone integrated service system, characterized in that the called terminal number is a new public telephone terminal number changed. According to claim 1, wherein the pre-movement destination network and the new destination network, The VoIP and public telephone integrated service system, characterized in that, when all operate in a VoIP network having a SIP proxy server, the pre-mobile destination network retransmits the outgoing call directly to a new destination terminal having the ported number. The method of claim 4 or 7, wherein the shifted number is, VoIP and public telephone integrated service system, characterized in that the called terminal number is a new SIP terminal address changed. In the Internet-based integrated network in which a VoIP network and a public telephone network are interworked, In the calling network, extracting the called terminal number in response to the call request of the calling terminal and transmitting the calling call to the called terminal through the first soft switch; Transmitting a call release information to the first soft switch when the called terminal number is changed to the moved number in response to the outgoing call in the pre-moving called network; Retransmitting the outgoing call to the changed address or number in response to the call release information in the outgoing network; And And in the new destination network, performing call connection routing between the originating terminal and the new destination terminal in response to the retransmitted outgoing call. 10. The method of claim 9, wherein the VoIP and public telephone integrated service method, In the pre-mobile destination network, in response to the outgoing call, when the called terminal number is changed to a transferred number, the method further includes directly retransmitting the outgoing call to a new called terminal having the moved number. VoIP and payphone integrated service methods made with. The method of claim 10, wherein the source network, VoIP and public telephone integrated service method operating in a public telephone network having a PSTN switch. The method of claim 10, wherein the source network, A VoIP and public telephone integrated service method operating in a VoIP network having a SIP proxy server. The method of claim 10, wherein the destination network before the movement, When the new destination network is operated on a VoIP network having a SIP proxy server, the before-mobile destination network includes the ported number via a second softswitch. Transmitting release information, and the originating network retransmits the outgoing call to a new destination terminal having the ported number in response to the call release information. The method of claim 10, wherein the destination network before the movement, When the new destination network operates on a public telephone network having a PSTN exchange, and the before-mobile destination network includes the ported number through a SIP proxy server. And the originating network retransmits the originating call to a new called terminal having the ported number in response to the call release information. The method of claim 14, wherein the shifted number is, VoIP and public telephone integrated service method, characterized in that the called terminal number is a new public telephone terminal number changed. 11. The method of claim 10, wherein the pre-movement destination network and the new destination network, And all of them operate in a VoIP network having a SIP proxy server, wherein the before-mobile destination network retransmits the outgoing call directly to a new destination terminal having the ported number. The method of claim 13 or 16, wherein the shifted number is, VoIP and public telephone integrated service method, characterized in that the called terminal number is a new SIP terminal address.